commit 6ac22e6b47f9a6ed70b0a376984b39b9a745dd94
author henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Aug 11 21:06:30 2014 +0000
committer henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Aug 11 21:06:30 2014 +0000
tree 4bcd4e92220d1376dd9c0db0afac1be7371a0ee3
parent 820f8e9ca7dddeef909cadb9411264942fba9dcb

Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798

R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/test/RTPFile.cc

diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.cc b/webrtc/modules/audio_coding/main/test/RTPFile.cc
index 50aee31..b886bde 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.cc
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.cc
@@ -234,10 +234,10 @@
     return 0;
   }
   if (payloadSize < (lengthBytes - 20)) {
-    return -1;
+    return 0;
   }
   if (lengthBytes < 20) {
-    return -1;
+    return 0;
   }
   lengthBytes -= 20;
   EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));